similar to: DTMF issue with TDM404

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition

At the same site, DTMF recognition is functioning badly, sometimes duplicating digits and sometimes totally missing others. We have checked already /proc/interrups, there is no interrupt being shared. Our zaptel has .. span=1,1,0,ccs,hdb3 On zapata we have ... switchtype=euroisdn signalling=pri_cpe relaxdtmf=no (yes doesn't seem to help) -----Original Message----- From:

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

Hi List, I am using Asterisk 1.6.2.18. One strange problem come into my knowledge after using this version of asterisk. Without pressing any digits or key from my mobile, I am getting DTMF into my asterisk server. For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and

Hi, I am using a channel bank and zaptel hardware. I have a credit card machine on one of the channels that appears to be dialing "too soon" for asterisk, every complete number recognized by asterisk is missing the first 1-4 numbers. This is a serious problem for me, anyone have any ideas on whats going on? The pstn picks up on the dtmf tones just fine. I was able to get it to

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

> I don't see anything obviously wrong with your configs. > You don't want relaxdtmf. That can cause the problem, not fix it. POST 2 --> got no response Hi Eric, at the begining -> Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but

Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

Ah I see, in theory it's possible then. We don't have any IVRs or anything which requires key presses, there isn't even voicemail on this particular phone system so I don't think it will be too much of a problem. I've also updated the firmware on the Cisco phones that have had the issue, just to see if that solves the issue but as it's been going on for a while, I'm

Hi I have a problem with dtmf recognition an analog lines connected to Sangoma A200. The digits (in most cases the first one) are doubled and so my IVR is useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but nothing worked. I also noticed one thing it only happens during the background application: exten => s,1,Background(soundfile) exten => 111,1,Dial(SIP/111)

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Hi Tom, Thank you for your informative and helpful reply. I had considered using the relaxdtmf setting but held off this due to not using any physical connection hardware -Asterik uses both SIP in and out from an upstream provider (Gradwell.com). Is it still possible to set this when using SIP trunks only and not physical hardware? The box does have a Digium ISDN card but the ISDN is no longer

Hi, I have Asterisk-1.2.18 install with FreePBX & more than 75 extnsion, daily I come accross an issue & try resolving them its either user learning curve or my ignorance. But, I dont know what to say regarding this issue. I have my Dial Plan for internal users to have a 3 Digit Extensions. So instance my Ext is 239 & someone dials the main #, its gets the

> I don't see anything obviously wrong with your configs. > You don't want relaxdtmf. That can cause the problem, not fix it. Hi Eric, at the begining -> Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but my SNOMs working pretty well

Hi, I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

I am having some troubles with DTMF detection on zap channels when the remote caller is calling from a noisy cell phone. It is actually detecting multiple DTMF tones (usually 2 or 3) when only one is sent (i.e. I press '3' and Asterisk is detecting that as '333'.) I don't know the exact situation with regards to signal strength, etc, but it seems to happen more often from users